1. Field of the Invention
The present invention relates to a media-gateway controller for call set-up processing and a method therefor, and more particularly, to a media-gateway controller for call set-up processing when a codec used by a caller is different from that used by a callee, and a method therefor.
2. Description of the Related Art
A media-gateway controller generally links a packet network to an existing wired/wireless network. In other words, a media-gateway controller transmits a signal between different types of networks, processes a call, and controls a media-gateway such as an access gateway or a trunk gateway.
An access gateway is needed to connect a normal telephone user on a wired/wireless network such as a public switched telephone network (PSTN) to a packet network such as a Voice over Internet Protocol (VoIP) network or a Voice over Asynchronous Transfer Mode (VoATM) network. The access gateway converts voice data from a normal telephone to be suitable for the packet network.
A trunk gateway links a PSTN to a packet network such as a VoIP or VoATM network and allows a large amount of data from the PSTN to be transmitted to the packet network.
A media-gateway is a data transformer for data transmission between networks comforting to different standards. The media-gateway includes an access gateway and a trunk gateway.
A media-gateway control protocol (MEGACO) is used for communication between a media-gateway and a media-gateway controller in the Internet not in an existing PSTN or a wireless communication network. In the existing PSTN and wireless network, a call processing unit and a media device are physically mounted to a single apparatus. However, in the Internet, a media device is separated from a call processing unit according to standardization of the MEGACO.
In the conventional PSTN and wireless communication network, a predetermined codec is used in each network. Accordingly, a conventional media-gateway makes codec conversion between networks. The standard recommendation for conventional call setting is described in RFC 3261 SIP, RFC 3264 Offer/Answer SDP, RFC 2833 RTP Payload for DTMP digits, Telephony Tones and Telephony Signals, RFC 2327 SDP, RFC 3108 ATM SDP, RFC 1890 RTP Profile Payload Type, etc. of the Internet Engineering Task Force (IETF).
A procedure for transcoding, i.e., codec conversion, for people having speech or hearing defects is disclosed in “Transcoding Services Invocation in the Session Initiation Protocol” of the IETF (http://www.ieff.org/internet-drafts/draft-camarillo-sip-deaf-01.pdf). However, this procedure requires a change in call processing application software of a caller's or a callee's terminal and development of a transcoding server.
A structure for linking different wireless communication networks is disclosed in U.S. Pat. No. 6,314,108, in which transcoding is performed between terminals in different networks using different codecs on the premise that a single network uses a single codec. However, this transcoding cannot be employed when a network uses a different codec from an existing one.
A conventional media-gateway controller performing codec conversion between different networks cannot set a call when a caller and a callee in one network provide different codecs or when a caller or a callee uses a different codec from that of a network including the caller or the callee. Even if a call is set, voice communication is not accomplished.